1. Technical Field of the Invention
This invention relates generally to multimedia content transport, and more particularly to the preparation for transport and transport of such multimedia content.
2. Related Art
The broadcast of digitized audio/video information (multimedia content) is well known. Limited access communication networks such as cable television systems, satellite television systems, and direct broadcast television systems support delivery of digitized multimedia content via controlled transport medium. In the case of a cable modem system, a dedicated network that includes cable modem plant is carefully controlled by the cable system provider to ensure that the multimedia content is robustly delivered to subscribers' receivers. Likewise, with satellite television systems, dedicated wireless spectrum robustly carries the multi-media content to subscribers' receivers. Further, in direct broadcast television systems such as High Definition (HD) broadcast systems, dedicated wireless spectrum robustly delivers the multi-media content from a transmitting tower to receiving devices. Robust delivery, resulting in timely receipt of the multimedia content by a receiving device is critical for the quality of delivered video and audio.
Some of these limited access communication networks now support on-demand programming in which multimedia content is directed to one, or a relatively few number of receiving devices. The number of on-demand programs that can be serviced by each of these types of systems depends upon, among other things, the availability of data throughput between a multimedia source device and the one or more receiving devices. Generally, this on-demand programming is initiated by one or more subscribers and serviced only upon initiation.
Publicly accessible communication networks, e.g., Local Area Networks (LANs), Wireless Local Area Networks (WLANs), Wide Area Networks (WANs), Wireless Wide Area Networks (WWANs), and cellular telephone networks, have evolved to the point where they now are capable of providing data rates sufficient to service streamed multimedia content. The format of the streamed multimedia content is similar/same as that that is serviced by the limited access networks, e.g., cable networks, satellite networks. However, each of these communication networks is shared by many users that compete for available data throughput. Resultantly, data packets carrying the streamed multimedia content are typically not given preferential treatment by these networks.
Generally, streamed multimedia content is formed/created by a first electronic device, e.g., web server, transmitted across one or more commutation networks, and received and processed by a second electronic device, e.g., Internet Protocol (IP) television, personal computer, laptop computer, cellular telephone, WLAN device, or WWAN device. In creating the multimedia content, the first electronic device obtains/retrieves multimedia content from a video camera or from a storage device, for example, and encodes the multimedia content to create encoded audio and video frames according to a standard format, e.g., MPEG-2 format. The audio and video frames are placed into data packets that are sequentially transmitted from the first electronic device onto a servicing communication network, the data packets addressed to one or more second electronic device(s). One or more communication networks carry the data packets to the second electronic device. The second electronic device receives the data packets, reorders the data packets, if required, and extracts the audio and video frames from the data packets. A decoder of the second electronic device receives the data packets and reconstructs a program clock based upon Program Clock References (PCRs) contained in the transport packets. The decoder then uses the reconstructed clock to decode the audio and video frames to produce audio and video data. The second electronic device then stores the video/audio data and/or presents the video/audio data to a user via a user interface.
To be compliant to the MPEG-2 Systems specification, PCRs are required to be inserted at least every 100 milliseconds in a MPEG-2 transport stream. In order for the second electronic device to accurately reconstruct the program clock, e.g., at 27 MHz, incoming transport packets must arrive at the decoder of the second electronic device with jitter that is less than approximately 1-2 milliseconds/30 parts-per-million (PPM). Data packets that are transported by communication networks such as the Internet, WANs, LANs, WWANs, WLANs, and/or cellular networks, for example, using IP addressing, for example, may travel via differing routes across one or more communication networks and arrive with various transmission latencies. In many operations, the data packets carrying timestamps arrive with significant jitter, sometimes approaching 200-400 parts-per-million jitter. With this large jitter, the receiving decoder may be unable to recreate the program clock from the received data packets.
Further, even if the decoder is able to recreate the program clock, the recreated program clock may have a significantly different frequency than the program clock of the encoding first device. Such differences in the recreated program clock of the decoder of the second electronic device as compared to the program clock of the encoder of the first electronic device results in buffer overflow or underflow at the second electronic device. Buffer overflow causes some of the incoming data to be purged and not buffered resulting in lost data and poor audio or video quality. Buffer underflow causes starvation of the decoder also resulting in poor audio or video quality. Further, even though the jitter in the transport stream may be relatively low, latency of transport may vary over time. Such is often the case with wireless networks that transport streamed multimedia content. In such cases, even though a program clock may be reconstructed from the PCRs that are received, data buffer underflow/overflow may also occur.
Thus, a need exists for a streamed transport system that operates satisfactorily with a high jitter transport stream, e.g. the Internet, and that produces video and audio output of high quality. Further limitations and disadvantages of conventional and traditional approaches will become apparent to one of skill in the art, through comparison of such systems with some aspects of the present invention as set forth in the remainder of the present application with reference to the drawings.